Impossible de passer un appel sortant

Bonjour à tous,

Alors voici dans quelle situation je me trouve.
Je suis client Sfr, j'ai pour projet de monter un petit Ipbx sur mon Raspberry et de le coupler à mon compte SFRLibertalk.
Pour le moment, je réalise mes tests sur sur une machine virtuel Debian avec asterisk 11.13.1.

Voici mon fichier Sip.conf :

Code:

---

[general]
register => +3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org:PASSWORD:NDI024316XXXX.LIBERTALK@sfr.fr@internet.p-cscf.sfr.net:5064~3600
allowguest=no
contactdeny=0.0.0.0/0.0.0.0
contactpermit=91.68.1.28/255.255.255.255 ; internet.p-cscf.sfr.net
contactpermit=192.168.1.0/255.255.255.0 ; mes réseaux privés
contactpermit=192.168.2.0/255.255.255.0 ;
alwaysauthreject=yes
media_address=78.XXX.58.XXX

[sfr-out]
type=peer
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
fromuser=+3399024316XXXX
defaultuser=NDI024316XXXX.LIBERTALK@sfr.fr
host=internet.p-cscf.sfr.net
insecure=invite,port
remotesecret=PASSWORD
auth=NDI024316XXXX.LIBERTALK@sfr.fr:PASSWORD@ims.mnc010.mcc208.3gppnetwork.org
outboundproxy=internet.p-cscf.sfr.net:5064
canreinvite=no

[sfr-in]
type=friend
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
host=internet.p-cscf.sfr.net
insecure=invite,port
context=from-sfr
port=5064
canreinvite=no

---

A ce jour, les appels internes fonctionnes, les appels entrants fonctionne aussi mais les appels sortants ne fonctionne pas.

Voici a quoi ressemble ma trace Sip d'un appel sortant :

Code:

---

SIP Debugging enabled

<--- SIP read from UDP:192.168.1.48:56563 --->

<------------->
[Jun 13 06:14:16] NOTICE[1808]: chan_sip.c:14995 sip_reregister:    -- Re-registration for  +3399024316XXXX@internet.p-cscf.sfr.net
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.68.1.28:5064:
REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK0545cdaa
Max-Forwards: 70
From: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as29a76624
To: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 3311dc0f34302aaa590258ab43bcb6d7@127.0.1.1
CSeq: 140 REGISTER
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="NDI024316XXXX.LIBERTALK@sfr.fr", realm="sfr.fr", algorithm=MD5, uri="sip:ims.mnc010.mcc208.3gppnetwork.org", nonce="b7c9036dbf3054ae577ac3f2a940e9703dc8f84c1608", response="0b59e9b9727f80c6eedd8c6c5df71170", opaque="ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRk5WGBkpdio3JnZyZiAnOGI-KD1-PzcnbmBmbmg_", qop=auth, cnonce="1cde1272", nc=00000002
Expires: 3600
Contact: <sip:s@192.168.1.89:5060>
Content-Length: 0


---

<--- SIP read from UDP:91.68.1.28:5064 --->
SIP/2.0 401 Unauthorized
Call-ID: 3311dc0f34302aaa590258ab43bcb6d7@127.0.1.1
Via: SIP/2.0/UDP 192.168.1.89:5060;received=78.XXX.58.XXX;branch=z9hG4bK0545cdaa
To: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=54d3a600-577ac4651563c85f
From: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as29a76624
CSeq: 140 REGISTER
Date: Mon, 04 Jul 2016 20:17:41 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
WWW-Authenticate: Digest realm="sfr.fr", nonce="b7c9036dbf3054ae577ac464a940e9703dc8f84c1608", opaque="ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRk5WGBkpdio3JnZyZiAnOGI-KD1-PzcnbmBmbmg_", stale=true, algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name internet.p-cscf.sfr.net
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.68.1.28:5064:
REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK2a67a3e7
Max-Forwards: 70
From: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as29a76624
To: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 3311dc0f34302aaa590258ab43bcb6d7@127.0.1.1
CSeq: 141 REGISTER
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="NDI024316XXXX.LIBERTALK@sfr.fr", realm="sfr.fr", algorithm=MD5, uri="sip:ims.mnc010.mcc208.3gppnetwork.org", nonce="b7c9036dbf3054ae577ac464a940e9703dc8f84c1608", response="0c6a25c9df01988012b9dbed243eed13", opaque="ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRk5WGBkpdio3JnZyZiAnOGI-KD1-PzcnbmBmbmg_", qop=auth, cnonce="21c24ca7", nc=00000001
Expires: 3600
Contact: <sip:s@192.168.1.89:5060>
Content-Length: 0


---

<--- SIP read from UDP:91.68.1.28:5064 --->
SIP/2.0 200 OK
Call-ID: 3311dc0f34302aaa590258ab43bcb6d7@127.0.1.1
Via: SIP/2.0/UDP 192.168.1.89:5060;received=78.XXX.58.XXX;branch=z9hG4bK2a67a3e7
To: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=54d3a600-577ac4651ad81c7e
From: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as29a76624
CSeq: 141 REGISTER
Allow-Events: reg
Contact: <sip:s@192.168.1.89:5060;transport=udp>;expires=3173
Date: Mon, 04 Jul 2016 20:17:41 GMT
Path: <sip:pcgw-0007.imsgroup0-003.ach4isc06.ims.sfr.net:5064;lr;ottag=ue_term;bidx=704717;access-type=ADSL>
P-Associated-URI: <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>
P-Associated-URI: <tel:+3399024316XXXX>
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
[Jun 13 06:14:16] NOTICE[1808]: chan_sip.c:23511 handle_response_register: Outbound Registration: Expiry for internet.p-cscf.sfr.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '3311dc0f34302aaa590258ab43bcb6d7@127.0.1.1' Method: REGISTER

<--- SIP read from UDP:192.168.1.48:56563 --->

<------------->

<--- SIP read from UDP:192.168.1.48:56563 --->
INVITE sip:062820XXXX@192.168.1.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:56563;rport;branch=z9hG4bKPjQRH897xLzYoNeJVQy.7r-3zt4pRpQDXo
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>
Contact: "Quentin PRXXXX" <sip:100@192.168.1.48:56563;ob>
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28254 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.1.7
Content-Type: application/sdp
Content-Length: 482

v=0
o=- 3676652280 3676652280 IN IP4 192.168.1.48
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
c=IN IP4 192.168.1.48
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.1.48
a=sendrecv
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (13 headers 22 lines) ---
Sending to 192.168.1.48:56563 (no NAT)
Sending to 192.168.1.48:56563 (no NAT)
Using INVITE request as basis request - cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
Found peer '100' for '100' from 192.168.1.48:56563

<--- Reliably Transmitting (no NAT) to 192.168.1.48:56563 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.48:56563;branch=z9hG4bKPjQRH897xLzYoNeJVQy.7r-3zt4pRpQDXo;received=192.168.1.48;rport=56563
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>;tag=as2e98c452
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28254 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0976aa86"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.48:56563 --->
ACK sip:062820XXXX@192.168.1.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:56563;rport;branch=z9hG4bKPjQRH897xLzYoNeJVQy.7r-3zt4pRpQDXo
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>;tag=as2e98c452
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28254 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.48:56563 --->
INVITE sip:062820XXXX@192.168.1.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:56563;rport;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>
Contact: "Quentin PRXXXX" <sip:100@192.168.1.48:56563;ob>
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.1.7
Authorization: Digest username="100", realm="asterisk", nonce="0976aa86", uri="sip:062820XXXX@192.168.1.89", response="5a83d095aa61380a6324af027db59d98", algorithm=MD5
Content-Type: application/sdp
Content-Length: 482

---

Suite de la trace :

Code:

---


<------------->
--- (14 headers 22 lines) ---
Sending to 192.168.1.48:56563 (no NAT)
Using INVITE request as basis request - cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
Found peer '100' for '100' from 192.168.1.48:56563
---
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.48:4000
Looking for 062820XXXX in appart (domain 192.168.1.89)
list_route: hop: <sip:100@192.168.1.48:56563;ob>

<--- Transmitting (no NAT) to 192.168.1.48:56563 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.48:56563;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY;received=192.168.1.48;rport=56563
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:062820XXXX@192.168.1.89:5060>
Content-Length: 0


<------------>
Audio is at 17004
Adding codec 100003 (ulaw) to SDP
,,,
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

---
Retransmitting #1 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

---
Retransmitting #2 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

---
Retransmitting #3 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

---

<--- SIP read from UDP:192.168.1.48:56563 --->

<------------->
Retransmitting #4 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

---
Retransmitting #5 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295
---

<--- SIP read from UDP:192.168.1.48:56563 --->
CANCEL sip:062820XXXX@192.168.1.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:56563;rport;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 CANCEL
User-Agent: Telephone 1.1.7
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.1.48:56563 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.1.48:56563 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.48:56563;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY;received=192.168.1.48;rport=56563
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>;tag=as72e83ee2
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.1.48:56563 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.48:56563;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY;received=192.168.1.48;rport=56563
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>;tag=as72e83ee2
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 CANCEL
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.48:56563 --->
ACK sip:062820XXXX@192.168.1.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.48:56563;rport;branch=z9hG4bKPjJzdyOoNy3pFR9-HeP7ozh7qV0vVAUCiY
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:100@192.168.1.89>;tag=vsTgcBkpVoQb6jymAEdN4pGwdSNO5GvB
To: <sip:062820XXXX@192.168.1.89>;tag=as72e83ee2
Call-ID: cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz
CSeq: 28255 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org' in 32000 ms (Method: INVITE)
Really destroying SIP dialog 'cQ8HbNBWUZVBBos6obplJs6x9CZia6Xz' Method: ACK

<--- SIP read from UDP:192.168.1.48:56563 --->

<------------->
Retransmitting #6 (no NAT) to 91.68.1.28:5064:
INVITE sip:062820XXXX@internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK32357084
Max-Forwards: 70
From: "Quentin PRXXXX" <sip:+3399024316XXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=as2ba7cfd0
To: <sip:062820XXXX@internet.p-cscf.sfr.net>
Contact: <sip:+3399024316XXXX@192.168.1.89:5060>
Call-ID: 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 13 Jun 2016 04:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

[Jun 13 06:15:03] WARNING[1808]: chan_sip.c:4028 retrans_pkt: Retransmission timeout reached on transmission 2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '2bef4c72658769477c19e0a33cf0fdb1@ims.mnc010.mcc208.3gppnetwork.org' Method: INVITE
SIP Debugging Disabled

---

D'avance merci si vous pouvez m'aider.

Quentin

Bonjour à tous,

Personne ne peut m'aider ?

problème réseau, des paquets se perdent:
Retransmitting #1 (no NAT) to 91.68.1.28:5064:

Citation:

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Envoyé par jean

problème réseau, des paquets se perdent:
Retransmitting #1 (no NAT) to 91.68.1.28:5064:

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quelle configuration reseau vous utilisez?

le serveur asterisk derriere la box sfr?

je suis comme vous coincé. j'y y etais arrivé il me semble en reroutant tous les flux externe vers le serveur asterisk. mode DMZ. le problème c'est qu'il n'y a plus rien d'autre qui marche dela box. plus de tele etc...... pas tres waf!!!

mes appels internes marchent nickel mais impossible de sortir ni d'entrer... j'y ai passé des heures.

les ports 5060 sont reservés par la box. aucun moyen d'utiliser ces ports de l'exterieur et de les router sur le serveur asterisk dans le LAN.

je n'aide pas mais je partage le meme problème.

-Olivier

il n'est normalement pas utile, voire dangereux, de router le port publique 5060 vers son serveur asterisk. Je te recommande de lire le lien dans ma signature sur le nat, ca devrait éclairer

J.